similar to: DTMF digits lost on TDM400 card

Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors.

Apologies if this is an old question; I've searched the list and the wiki but have not been able to find a definitive answer. I have an Aastra 480i phone registered with * 1.2.4; I want to generate UK ringback tones when the handset dials another internal extension. On my Zap channels, I have this in place by editing /etc/zaptel.conf; however I've had no luck with the Sip handset (I have

dude it's gotta be something with your system. Im using same setup at home with a TDM22 with no probs. Asterisk@Home 1.5, Compaq Deskpro EN P3 500, cordless phones. You did a ton more than I did, I basically plugged everything in and installed a@h.Worked first try. Hate to say it, but would it be possible to try a different system? Even an old system as long as it is PCI 2.2. The big clue

Folks, I have a TDM400 with an FXS module. I'm having some trouble ringing some phones attached to the device. I have a real el cheapo handset that cost me about 9UKP. It works fine, I plug it into the Wildcard and call the channel and it rings. I have tried plugging in my BT Diverse 3016 cordless phone but it will not ring, although I can call out from it. I have also tried my GPO 332L

Hi, I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO. I'm using a VTech cordless that makes three short beeps when someone another extension is picked up, presumably this lets you know if someone is trying to listen in.. Everything works, except the VTech now makes the three beeps everytime you try to use it, even if another extension is in use. It seems as though the

I'd planning on install Asterisk on a hosted Linux box we're setting up. The hosting provider that seems to offer the best deal can install either Debian 3.1 or SUSE 9.x or 10.0 in either 32 or 64 bit editions (running on AMD 64 bit). My experience has been gained on RedHat to date, but I do have some SUSE experience. I assume I'll have no problems running Asterisk on SUSE, but

Hi all, I've been trying to get SMS operational on my Asterisk box, which has a TDM400P card with a pair of FXO interfaces configured (ZAP/1 & ZAP/2). I've not had luck with either of my lines, after issuing the command "smsq --motx-channel=ZAP/1/1709400X 00000 register". I see the following output in my Asterisk console: -- Attempting call on ZAP/1/17094009 for

I am surprised you needed to turn the rxgain down so much, usually it is just the opposite. I experienced the same problem you did when my txgain was too low. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May

Hello, I'm not detecting caller Id through DTMF with my TDM400 card and don't know how to put this to work. Could someone please help me ? How should I configure * to achieve this? Thanks, Fernando

Hello, I am trying to make my Asterisk to recognize CallerID from incoming PSTN calls. I am using a TDM400. CallerID in my country's PSTN is based on DTMF. Some information from my configuration files: /etc/zaptel.conf: loadzone=nl defaultzone=nl /etc/asterisk/zapata.conf usecallerid=yes cidsignalling=dtmf cidstart=polarity As displayed on the console, my Asterisk cannot detect the event

Hi My card is working, but when I reboot the machine, most of the times it is not working, I get "ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6)" To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose "Ignore" and then it's

OK, I have a Asterisk @ Home 1.0.7 server with two Digium TDM400 cards, one 4 port FXO and one 4 port FXS. When I plug an analog cordless phone into the TDM40B card and setup a ZAP extension, the phone rings in and you can answer just fine. The weird part is when you try to get dialtone from the cordless handset it rings all of the other extensions... One other thing to mention is that the

I'm having a big problem where digits dialed from certain cell phones are too short to be recognized by my asterisk server. I'm running AAH 2.8. Some cell phones don't allow the caller to hold down the digits and have the tones play as long as they hold them down for. They just play a short tone no matter how long you hold down the digits for. Has anyone run into this before, and if

Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com

I'm trying to install * 1.2.5 for the first time on SUSE 10, 64bit (I've previously used Fedora or Centos). After I installed the kernel sources and followed some instructions on Novell's web site to copy the Module.symvers.gz file across from /boot and do a make modules_prepare, I was successfully able to complete 'make linux26' in my /usr/src/zaptel folder. I'm still

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

hey all, I want to set up a phone network at home. I have to phone lines (one traditional line and another provided by my internet provider (voIP)). So what I would like to do is get a Digium box and connect my two lines to it and then use several cordless voip phones with the ability to transfert call from one phone to another. What kind of digium hardware i need to do this? also I will use

Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: "Thank you for calling xxxx". I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the "phpconfig for Asterisk PBX" page, and selected

-------------------------------------------------------------------------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using

I have a * box running our house and on one extension we are getting spurious DMTF tones during conversations. It only happens on one of the 3 FXS ports and it's the one w/ a cordless phone on it. At first I thought someone was being careless and just hitting a button on the other end of the line, but it's happening too much for that... Has anyone run into this before? -- -M There